Implement writing mic audio input data to file for debugging purposes.

content/browser/media/webrtc_internals.h

 // Copyright (c) 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef CONTENT_BROWSER_MEDIA_WEBRTC_INTERNALS_H_
 #define CONTENT_BROWSER_MEDIA_WEBRTC_INTERNALS_H_
 
 #include "base/containers/hash_tables.h"
 #include "base/gtest_prod_util.h"
 #include "base/lazy_instance.h"
@@ skipping 69 matching lines @@
                       const std::string& audio_constraints,
                       const std::string& video_constraints);
 
   // Methods for adding or removing WebRTCInternalsUIObserver.
   void AddObserver(WebRTCInternalsUIObserver *observer);
   void RemoveObserver(WebRTCInternalsUIObserver *observer);
 
   // Sends all update data to |observer|.
   void UpdateObserver(WebRTCInternalsUIObserver* observer);
 
-  // Enables or disables AEC dump (diagnostic echo canceller recording).
-  void EnableAecDump(content::WebContents* web_contents);
-  void DisableAecDump();
+  // Enables or disables diagnostic audio recordings for debugging purposes.
+  void EnableAudioDebugRecordings(content::WebContents* web_contents);
+  void DisableAudioDebugRecordings();
 
-  bool aec_dump_enabled() {
-    return aec_dump_enabled_;
-  }
-
-  base::FilePath aec_dump_file_path() {
-    return aec_dump_file_path_;
-  }
+  bool IsAudioDebugRecordingsEnabled() const;
+  const base::FilePath& GetAudioDebugRecordingsFilePath() const;
 
   void ResetForTesting();
 
  private:
   friend struct base::DefaultLazyInstanceTraits<WebRTCInternals>;
   FRIEND_TEST_ALL_PREFIXES(WebRtcAecDumpBrowserTest, CallWithAecDump);
   FRIEND_TEST_ALL_PREFIXES(WebRtcAecDumpBrowserTest,
                            CallWithAecDumpEnabledThenDisabled);
   FRIEND_TEST_ALL_PREFIXES(WebRtcAecDumpBrowserTest, TwoCallsWithAecDump);
   FRIEND_TEST_ALL_PREFIXES(WebRtcInternalsTest,
@@ skipping 10 matching lines @@
   // ui::SelectFileDialog::Listener implementation.
   void FileSelected(const base::FilePath& path,
                     int index,
                     void* unused_params) override;
   void FileSelectionCanceled(void* params) override;
 
   // Called when a renderer exits (including crashes).
   void OnRendererExit(int render_process_id);
 
 #if defined(ENABLE_WEBRTC)
-  // Enables AEC dump on all render process hosts using |aec_dump_file_path_|.
-  void EnableAecDumpOnAllRenderProcessHosts();
+  // Enables diagnostic audio recordings on all render process hosts using
+  // |audio_debug_recordings_file_path_|.
+  void EnableAudioDebugRecordingsOnAllRenderProcessHosts();
 #endif
 
   // Called whenever an element is added to or removed from
   // |peer_connection_data_| to impose/release a block on suspending the current
   // application for power-saving.
   void CreateOrReleasePowerSaveBlocker();
 
   base::ObserverList<WebRTCInternalsUIObserver> observers_;
 
   // |peer_connection_data_| is a list containing all the PeerConnection
@@ skipping 17 matching lines @@
   // "rid" -- the renderer id.
   // "pid" -- proceddId of the renderer.
   // "origin" -- the security origin of the request.
   // "audio" -- the serialized audio constraints if audio is requested.
   // "video" -- the serialized video constraints if video is requested.
   base::ListValue get_user_media_requests_;
 
   // For managing select file dialog.
   scoped_refptr<ui::SelectFileDialog> select_file_dialog_;
 
-  // AEC dump (diagnostic echo canceller recording) state.
-  bool aec_dump_enabled_;
-  base::FilePath aec_dump_file_path_;
+  // Diagnostic audio recording state.
+  bool audio_debug_recordings_;
+  base::FilePath audio_debug_recordings_file_path_;
 
   // While |peer_connection_data_| is non-empty, hold an instance of
   // PowerSaveBlocker.  This prevents the application from being suspended while
   // remoting.
   scoped_ptr<PowerSaveBlocker> power_save_blocker_;
 
   // Set of render process hosts that |this| is registered as an observer on.
   base::hash_set<int> render_process_id_set_;
 };
 
 }  // namespace content
 
 #endif  // CONTENT_BROWSER_MEDIA_WEBRTC_INTERNALS_H_